Jan 16, 2009

DTMF Relay : RTP-NTE vs SIP INFO vs SIP NOTIFY

DTMF tones are the tones that are generated when a telephone key is pressed on a touchtone phone. Sometimes the called endpoint needs to hear those tones, such as when you enter digits during the call in response to a menu. Low-bandwidth codecs can distort the sound, however. DTMF relay allows that tone information to be reliably passed from one endpoint to the other. By default, SIP uses in-band signaling, sending the DTMF information in the voice stream. However, you can configure it to use RTP-NTE, SIP INFO messages, SIP NOTIFY messages, or KPML for transmitting DTMF tone information.

RTP-NTE is an in-band DTMF relay method, which uses RTP Named Telephony Event (NTE) packets to carry DTMF information instead of voice.
If RTP-NTE is configured, SDP is used to negotiate the payload type value for NTE packets and the events that will be sent using NTE.

RTP-NTE can cause problems communicating with SCCP phones, which use only out-of-band DTMF relay. In a CallManager 4.x network with SCCP phones, you must provision an MTP for calls that traverse the SIP trunk. This MTP translates between in-band and out-of-band DTMF signals. You must configure a separate MTP for each side of the SIP trunk. You can do this MTP in hardware, or in software on CallManager.

Cisco has two out-of-band procedures for DTMF relay. One uses SIP INFO methods, and the other uses SIP NOTIFY methods.
The SIP INFO method sends DTMF digits in INFO messages. It is always enabled. When a gateway receives an INFO message containing DTMF relay information, it sends the corresponding tone.

NOTIFY-based out-of-band DTMF relay is negotiated by including a Call-Info field in the SIP INVITE and response messages. This field indicates an ability to use NOTIFY for DTMF tones and the duration of each tone in milliseconds. Using this method can help SIP gateways interoperate with Skinny phones. You can also use it for analog phones that are connected to Foreign Exchange Station (FXS) ports on the gateway.

When a DTMF tone is generated, the gateway sends a NOTIFY message to the terminating gateway. When that gateways receives the NOTIFY, it responds with SIP 200 OK and plays the DTMF tone. To configure the DTMF relay type, use the dtmf-relay command in dial-peer configuration mode. To optionally configure the interval between NOTIFY messages for a single DTMF event, use the notify telephone-event max-duration milliseconds command in SIP UA configuration mode. The default is 2000 msec; the lowest value between two SIP peers is the one chosen.

KPML is another way for SIP phones to send dialed-digit information. Like SIP-NOTIFY, KPML uses a NOTIFY message to transmit each digit.

Media Termination Point (MTP)

A Media Termination Point (MTP) software device allows Cisco CallManager to relay calls that are routed through SIP or H.323 endpoints or gateways.

MTP, a Cisco software application, installs on a server during the software installation process. You must activate and start the Cisco IP Voice Media Streaming App service on the server on which you configure the MTP device. For information on activating and starting services, refer to the Cisco CallManager Serviceability Administration Guide.

Each MTP device that is defined in the database registers with the Media Resource Manager (MRM). The MRM keeps track of the total available MTP devices in the system and of which devices have available resources.

During resource reservation, the MRM determines the number of resources and identifies the media resource type (in this case, the MTP) and the location of the registered MTP device. The MRM updates its share resource table with the registration information and propagates the registered information to the other Cisco CallManagers within the cluster.

The MTP and transcoder can register with the same Cisco CallManager. See the "Transcoder Configuration" section on page 30-1 for more information.

Each MTP receives a list of Cisco CallManagers, in priority order, to which it should attempt to register. Each MTP can register with only one Cisco CallManager at a time.

Jan 15, 2009

Super Group 3(SG3)

Super Group 3 (SG3) is a standard of fax machines that support speeds of up to 33.6 kbps through V.34 half duplex (HD) modulation and V.8 signaling.

Prior to Cisco IOS Release 12.4(4)T, SG3 fax machines could interoperate only over T.38 fax-relay and Cisco fax-relay networks with G3 fax machines, not with other SG3 fax machines, unless the fax machines were specifically configured to work at slower speeds or were configured for modem pass-through. The use of SG3 V.8 fax CM message suppression provides a gateway-controlled solution that enables SG3 fax machines to scale down without end-user interaction and without the extra bandwidth required by modem pass-through.

SG3 V.8 fax CM message suppression allows SG3 fax machines to interoperate over a fax-relay network at G3 speeds by blocking the SG3 V.8 CM message, or fax tone, from reaching the called fax machine. This causes the called fax machine to time out on the ANSam tone and scale down to G3 speeds by initiating V.21 negotiations.

Error Correction Mode(ECM)

Error correction mode (ECM) is an optional transmission mode built into Class 1 fax machines or fax modems. ECM automatically detects and corrects errors in the fax transmission process that are sometimes caused by telephone line noise. The page data is divided into what is known as Octets (small blocks of data).

Once the receiver has received all the Octets it examines them (using check-sums) and then advises the transmitting fax of any Octets that are in error. The transmitter then need only resend the blocks in error rather than the whole page. This generally means an ECM coded fax will be more likely to succeed on a noisy line. ECM is the norm rather than the exception. Some fax machines have the capability to enable or disable this function.

Jan 14, 2009

Changes to CCIE Lab and Written Exam Question Format and Scoring

看來大陸同胞們無法再飛到其他國家去考CCIE Lab來逃避CCIE Lab口試了,未來的CCIE Lab全面加上了口試至少可以保證CCIE的產出有一定的水平,不過這對東方人來說也許只是增加新的題庫而已,呵,事實上北京那邊的口試題目已經被大陸同胞們整理的差不多了,只要是有制式題目及答案的話,應該還是"上有政策,下有對策"。

Effective February 1, 2009, Cisco will introduce a new type of question format to CCIE Routing and Switching lab exams.
In addition to the live configuration scenarios, candidates will be asked a series of four or five open-ended questions, drawn from a pool of questions based on the material covered on the lab blueprint. No new topics are being added. The exams are not been increased in difficulty and the well-prepared candidate should have no trouble answering the questions. The length of the exam will remain eight hours. Candidates will need to achieve a passing score on both the open-ended questions and the lab portion in order to pass the lab and become certified. Other CCIE tracks will change over the next year, with exact dates announced in advance.

Effective February 17th, 2009, candidates will also see two other changes in CCIE written exams.
First, candidates will now be required to answer each question before moving on to the next question; candidates will no longer be allowed to skip a question and come back to it at a later time. Second, there will be an update to the score report. The overall exam score and the exam passing score will now be reported as a scaled score, on a scale from 300-1000. This change will not affect the difficulty of the current set of exams and will assure CCIE written exams will be consistent with Cisco’s other career certification exams.